Hello All:Would love some information on VBR and ABR MP3's. When I started My Digital Music Collection a few years ago I started with 256 and 320 CBR MP3's. I usually ripped my music with WMP. Last year, I got rid of WMP and switched over to another media player. It was some free player which was called fre:ac which had a newer version of lame. I still used CBR's and I found them to sound better then the songs ripped by my old version of WMP, but I started messing around with other formats. I found that wma's (non protected) at 256 to 320 outshined my CBR MP3's at the same bitrate, so I started using the WMA's. This led to a problem. Even though my three mp3 players all suport wma's at 320, the wma's hit the battery hard. The wma's seem to suck about 50% more battery life then the MP3's did. (and I still had mp3 on my player, but majority of my music was wma.) So I did some research and found that VBR or ABR mp3's are the way to go, giving you better quality. I tried this and I really like the way they sound. I also downloaded media monkey and use this to rip my CD's. MM was a highly recommended media player and even though I dont know what version of lame it is using, it seems to do the job fairly well. So here comes my questions and sorry about the long introduction. I need some advice for the settings. I have played around with many and I have found that using the ~245 setting with no advanced settings seem to do the trick. I listen to metal, classic rock+pop/oldies, electroinica and jazz and I usually end up with a bit rate of 280 for metal and electronica and 255 for jazz and classic rock. I would love some advice some of my other settings and would love some opinions of my settings. Hopefully you will have some fun with this as I know some of you are audiophiles and would love your opinions. I am trying to get the best quality of my music with the lowest Bit rate possible so the battery doesnt get eaten up. Here are my settings: 1.~245 (no advance settings) This seems pretty good as stated before, but without any setting to let you know what you set the lowest bitrate for, sometimes I have songs that drop down to an "average bit rate" of 240. This happened with John Coltrane and Miles Davis. Also with Traffic and Steely Dan the average was about 260. I thought this was a bit low for those types of music. I thought Jazz and Progressive Classic Rock such as Steely Dan,Yes, and Pink Floyd demanded a higher bit rate and even though the music would peak up in the 320's at times (hence the advantage of VBR) this seems kinda low for an average.2.~245 with advance settings of lowest bit rate of 256 and highest of 320. I have bit resoviour turned off. This setting is good as I get a higher avreage bitrate, usually 295-305. Pretty Close to the 320 mark I am sure that its just the same. Most all types of Music come out at around that Bit-rate. Big question about this is that if a song hits a passage of lets say 160, does the encoder still go that low since I set lowest to 256.?3. 300ABR-I use this when I get a file that has an average Bit rate that I think is to low with the other previous settings. I heard its almost like VBR, but I cant seem to find anything on its actual behaviors. With this I get a a average Bit rate of 290.Are the Bitrates I am choosing Overkill? What setting would you choose? I think that there might be a slight difference between a file playing at an average of 225 then 305 but it just might be like a placebo effect and once I know I am playing a song at a lower average bitrate it doesnt sound as good so my ears are helping me in my test..lol Does anyone hear a difference with large numbers of values in bitrates? I understand this is really difficult because so many people have different views. I know some people that say 192cbr is cd quality (i disagree, but if thats what your ears tell you then go for it) I just want to see what the majority has to say of where they like their average bit rate to be or what they think is needed to achieve highest quality. Love to hear from as many as possible

I'm not sure what you mean by determining quality. I read the rules of the forum and I am not understanding.I am so sorry I am new to this. Maybe these statements will help. My main goal was what experts and people think what the average bit rate should be or come to with certain types of music to achieve the highest quality (well the highest quailty that the MP3 can go, since MP3 is lossy) and smallest file size. Its hard for me to determine since I think my ears have a placebo effect. Its like once know what my MP3 is playing at, its like the one with the lower bit rate doesnt sound as good. I just dont know if this is because have become hard-headed and think a greater bit rate will be better even if that bit rate is only a difference of maybe 30 or 40kbps. I seached and seached this forum on these topic but I can't seem to find anyone that has asked such a spefic question about a VBR or ABR bitrate. Most of the post all refer to a CBR or a CBR vs a VBR but at what bitrate (nor have I read of anyone using ABR's in the 300bitrate range). If I hear from experts that have been dealing with VBR's for years (remember I am new to them) either by listening or by some sort of graph or tester (sorry i have no ideas what the names of these things are) This way I know what setting to choose, that I no longer have to worry about eating up to much space and perserving battery life with the best possible SQ. Hope this helps and that my posting is still allowed.

- Your original posting contained a few subjective opinions of the type that this codec of that bitrate was better than (“outshine” is quite a strong word) some competition. As long as those statements refers to sound quality (rather than e.g. to battery life), then they are per the terms of service not welcome on this forum unless backed by evidence (see the TOS) or infeasible to test (like “I had to rebuild my house after the fire, but at least I got rid of that annoying boom at 67 Hz” would be a statement way beyond your ability to back up by a listening test). Of course moderators will likely not knock you for stating “I tried to save space by going mono, but that sounded too annoying”.

- If you did transcode (i.e. re-encoding a lossy to a lossy), then it is known that there is a generation loss which can make artifacts audible at bitrates which would be transparent if you did encode from a lossless source. If you are looking for advice on what to do about your lossy files, the general answer is “keep them”.

- If you are looking for general advice on settings to choose to minimize the filesize for given level of annoying artifacts, then I'd say the broad consensus is that VBR should be your first shot. Not equally universally agreed upon, but still likely to be best advice: if you worry that e.g. LAME V0 isn't good enough, then just stay lossless.

- If you have a lossless archive, but need space-saving settings for portable use, then why not start low? If it sounds annoying, just overwrite.

As for lossy encoder settings for best quality, it's a common question, and one that doesn't have a general answer that's going to be right for everybody. You have to generate the answer yourself, by testing. If you conduct proper ABX tests, you'll probably find that your suspicions about the placebo effect are quite correct, and that generally you don't need even half the bitrate or specific format you think you do, at least for getting "best quality". Oh, and you have to define "quality". What is that, for you?

The idea behind lossy is that the audio data can be simplified in space-saving ways that will quantitatively change the output, but that won't affect our perception at all, or that will affect it as minimally and least-unpleasantly as possible. The point at which a given piece of lossy-encoded audio, played through your equipment and heard by your ears, is indistinguishable from the original is the point of "transparency". Any knob-twiddling you do to "increase the quality" beyond that point is not really increasing the quality at all, because it's already at maximum for you. And as you might expect, where it is for you is not where it is for everyone else, and can be affected by the choice of audio to encode, your hearing loss, background noise, encoder, settings, etc.

Determining the point of transparency requires conducting a series of tests where you compare the original to two different clips: one is an exact copy of the original, the other is your lossy test version. These must all be volume-matched. In the test, you're forced to say which is which, even if you can't tell the difference. Do this enough times, and you'll have a meaningful score... 50% wrong answers (same as flipping a coin) means you couldn't tell the difference, less than 5% wrong means we won't argue with you anymore. foobar2000 is really good for ABX testing if you're doing it on your PC, and I believe there are ABX apps for smartphones, but it's going to be difficult to do a proper test with, say, an iPod Mini.

Its hard for me to determine since I think my ears have a placebo effect.

The first link I provided tells you how to avoid this very real and negatively influential phenomenon. Please read it.

In the second link please read about rule #8, in the first post as well as in the later post that describes the rule in more detail.

At this point I think you can safely discard any of the conclusions you've presented about different compression settings and how they affect sound quality.

I re-read the first link you sent me and very interesting stuff! It lamemans terms if I am perceiving it correctly it is almost like the human ear has no way of telling us about compression. It almost seems like it needs to be done with an abx test. "Science," as Thomas Dolby said. Since you seem like an expert on this field, does this mean that the only way to determine this is to break down every single song with this this test. It sounds like this would take decades with those with a large music collection. Greynol are you allowed to give your opinion on a matter such as Bitrate. I have no intentions of starting arguements as I can see you dont like those. I have read many post and once it seems if someone makes a statement and has no way to back it up, you will Greynol them..lol...I would love your opinion even if it isn't scientific on what bit rate that you like to stay at. Once again I dont know if you are allowed to do this according to the rules of the forums, but this seems like one of the better forums on the web when dealing with mp3's. I have so many lame post on sites where people argue about 192 vs 320. I am not interested in that garbage. I am interested in what the experts use when they are using MP3's if you even use them at all. Do the experts or yourself find any difference with lossy vs lossless without the help of technology. If you are allowed to make a statement that is pure opinion and what you use in the real world I would love to hear it. I figure take the advice of those that know, not because some extemist says 320 or die

As for lossy encoder settings for best quality, it's a common question, and one that doesn't have a general answer that's going to be right for everybody. You have to generate the answer yourself, by testing. If you conduct proper ABX tests, you'll probably find that your suspicions about the placebo effect are quite correct, and that generally you don't need even half the bitrate or specific format you think you do, at least for getting "best quality". Oh, and you have to define "quality". What is that, for you?

The idea behind lossy is that the audio data can be simplified in space-saving ways that will quantitatively change the output, but that won't affect our perception at all, or that will affect it as minimally and least-unpleasantly as possible. The point at which a given piece of lossy-encoded audio, played through your equipment and heard by your ears, is indistinguishable from the original is the point of "transparency". Any knob-twiddling you do to "increase the quality" beyond that point is not really increasing the quality at all, because it's already at maximum for you. And as you might expect, where it is for you is not where it is for everyone else, and can be affected by the choice of audio to encode, your hearing loss, background noise, encoder, settings, etc.

Determining the point of transparency requires conducting a series of tests where you compare the original to two different clips: one is an exact copy of the original, the other is your lossy test version. These must all be volume-matched. In the test, you're forced to say which is which, even if you can't tell the difference. Do this enough times, and you'll have a meaningful score... 50% wrong answers (same as flipping a coin) means you couldn't tell the difference, less than 5% wrong means we won't argue with you anymore. foobar2000 is really good for ABX testing if you're doing it on your PC, and I believe there are ABX apps for smartphones, but it's going to be difficult to do a proper test with, say, an iPod Mini.

I am so sorry with all my rambling about quality and I never stated my meaning. Quality meaning staying as close to the original musical source (CD or WAVE File) as possible even if it is lossy.

- Your original posting contained a few subjective opinions of the type that this codec of that bitrate was better than (“outshine” is quite a strong word) some competition. As long as those statements refers to sound quality (rather than e.g. to battery life), then they are per the terms of service not welcome on this forum unless backed by evidence (see the TOS) or infeasible to test (like “I had to rebuild my house after the fire, but at least I got rid of that annoying boom at 67 Hz” would be a statement way beyond your ability to back up by a listening test). Of course moderators will likely not knock you for stating “I tried to save space by going mono, but that sounded too annoying”.

- If you did transcode (i.e. re-encoding a lossy to a lossy), then it is known that there is a generation loss which can make artifacts audible at bitrates which would be transparent if you did encode from a lossless source. If you are looking for advice on what to do about your lossy files, the general answer is “keep them”.

- If you are looking for general advice on settings to choose to minimize the filesize for given level of annoying artifacts, then I'd say the broad consensus is that VBR should be your first shot. Not equally universally agreed upon, but still likely to be best advice: if you worry that e.g. LAME V0 isn't good enough, then just stay lossless.

- If you have a lossless archive, but need space-saving settings for portable use, then why not start low? If it sounds annoying, just overwrite.

Yes it was an opinion of mine with the word outshined. That was then though..lol..That was my hardhead days when It was 320 or die..I have tried to be open to the experts at what is the best way to go to achieve the quality of the orginal CD to the best ability by going lossy (my mp3 players dont support wav, flaq) I figure what is the point of using high bit rates unless they are needed. Eats up space and I think it eats up battery life. Mine Mp3 player did with wma's but thats only on my player. I cant tell you it there is any real evidence of this expect word of mouth from others

The most common answer to the question, "which lossy settings should I choose in order not to waste space?" is the following:

Pick a few tracks that are representative of your collection and encode them at a relatively low bitrate ~100kbits (for mp3) and see if you can ABX them from the original lossless source using foobar2000's ABX comparator. Increase the bitrate until you can you can no longer distinguish a difference.

For my own tests, I was able to ABX Lame at -V5 and sometimes -V4, so I ended up choosing -V3 in order to give me some margin which I felt that I could afford. Someday I might try this with AAC since it seems likely that it will save me more space, but as of yet I haven't seen the need.

This post has been edited by greynol: Apr 26 2013, 23:01

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Breath is found in waveform and spectral plots;DR figures too, of course.

a) With your bitrate settings chance is very low that you are able to hear differences. That's why greynol asked for how you determined quality.

b) The right way to check whether or not there is an audible difference to the original is by doing an ABX (blind listening) test. Probably the easiest way to do this is to use the ABX tool which comes with foobar2000. foobar is also a great tool for doing audio conversion and things like that, or to just listen to the music.

c) Disabling bitreservoir is a bad idea. Bitreservoir always has a positive effect on quality.

d) There is a widespread misconception about bitrate. In an mp3 file there are two kinds of bitrates. The most important bitrate is that of the audio data. But the audio data are packaged into frames (containers for the audio data) which have their own bitrates (which corresponds to the size of these containers). Bitreservoir allows audio data to spread over several frames. That's why there is only a loose relation between audio data bitrate and frame bitrate. When you demand Lame to keep bitrate at 256 kbps or above this is a specification for the frame bitrate, not the audio data bitrate. When you additionally disable bitreservoir you don't change audio data at all. You just use bigger containers for the same audio data. You have a lot of unused bits in your files.You can figure that out by using the mp3packer tool on your files. You'll get smaller files with exactly the same audio data (to ensure yourself that I'm not talking nonsense you can use foobar's bit-compare track tool).If you keep bitreservoir enabled this can have a positive quality effect when keeping (frame) bitrate at 256 kbps or above because this way the amount of data available in the bitreservoir will be a bit higher than when using the default settings. But you should use mp3packer afterwards to squeeze the many unused bits out of the file. And you should use 320 kbps as the general frame bitrate - when using mp3packer afterwards average bitrate won't increase.

e) With very high bitrate settings as you use them audible deviation from the original is very rare but it can happen. The essential question is how to deal with this situation, and you will find two groups of people here which often can't understand the other group's attitude. The probably major group feels like 'I don't care about rare events when quality isn't great. Usually quality is perfect, and when it isn't it's usually not annoying.' These people usually use -V2 or similar according to their personal attitude and needs. Sometimes they are additionally proud of going this way because they use their bits very effeciently and think it's a misuse of a lossy encoder to make use of it in a less efficient way.Compared to this group the other group is a bit paranoid. Audible issues even when they are rare give them an uncomfortable feeling. They want to be very much on the safe side. Usually they still care about file size, but to a very minor extent especially as these days sufficient storage space usually is available or can be made available even on portable devices at low price.I personally belong to the paranoid group, and you sound very much like you do too.My advice: use lame3100i, a functional extension. lame3100i has a minimum audio data bitrate feature for instance, something that you obviously want to use but can't get with standard Lame which just allows for defining the mnimum frame bitrate.

For those who belong in the first category, you can always re-encode* using more liberal settings once you encounter a problematic sample. Even if this ever happens (NB my use of italics) then you know you haven't wasted any bits by bloating your collection by an additional 40(?), 50(?), 60(?), 70(?)%.

(*) EDIT: There is no law that states you have to re-encode your entire collection just because you found a problematic sample. You can simply re-encode just the ones that are problematic.

This post has been edited by greynol: Jun 1 2013, 19:38

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Breath is found in waveform and spectral plots;DR figures too, of course.

ThomasG3rd, I can suggest for the metal part of your collection to encode with -V 0 (no other settings) and use latest LAME. This is based on my personal experience. For a long time I used the setting -V 4, since it was transparent for almost all my metal music. However, I have since found a couple of problem samples (with some tracks from Nightwish, Amon Amarth, Keep of Kalessin,...) where I could achieve transparency only by using -V 1 or sometimes even -V 0. That led me to use only -V 0 from then on, just to be on the sure side. I know I am mostly wasting space, but knowing that I am doing all I can to avoid getting some problematic metal song to encode with artifacts is more important to me than the space those mp3s use, since space became really cheap and is not the issue nowadays.I hope this helps...About the listening tests, one Nightwish track has it's own topic on this forum and I can post listening test from the other tracks if needed.

2.~245 with advance settings of lowest bit rate of 256 and highest of 320.

Why? Then you are crippling the VBR mode by only allowing it to use frames of two sizes.

QUOTE

I have bit resoviour turned off.

Why? Then you are crippling the VBR mode by not allowing it to divert bits from simpler frames to more complex ones.

QUOTE

This setting is good as I get a higher avreage bitrate, usually 295-305.

Of course you get higher mean bitrates. You have removed the encoder’s ability to use smaller frames when they are appropriate.

QUOTE

Pretty Close to the 320 mark I am sure that its just the same.

Of course it’s not “just the same”.

QUOTE

Big question about this is that if a song hits a passage of lets say 160, does the encoder still go that low since I set lowest to 256.?

How is this not obvious? Of course it cannot.

Did you use a dartboard to choose these settings? The defaults are the defaults for a reason, and I advise against changing them in any case, especially when the user does not seem actually to know what they’re doing.

Even if you were using sensible methods to increase the mean bitrate, that might be totally unnecessary if lower bitrates would equally provide perceptual transparency. I have to recommend further reading on both (1) LAMEs actual settings, rather than whatever oversimplified descriptions MediaMonkey presents in their stead, and (2) double-blind testing. Other users have already provided a lot of feedback, but you might want to start simpler, for example at the page about LAME on our wiki.

QUOTE (ThomasG3rd @ Apr 26 2013, 22:40)

I am so sorry with all my rambling about quality and I never stated my meaning. Quality meaning staying as close to the original musical source (CD or WAVE File) as possible even if it is lossy.

Talking of degrees of closeness to the source is irrelevant for lossy encoding (at least in terms of listening, which is what matters here). A lossily encoded file is either perceptually transparent or not. Beyond that, any increase in bitrate is a waste. Sure, there will always be rare exceptions that might not encode transparently at your chosen setting, but that doesnt justify encoding everything unnecessarily high.

Besides, it seems futile to me to try to rank which settings in LAME are best in that regard, as CBR and VBR have some fundamental differences in behaviour, so 320 kbps CBR may not be better than a high VBR setting (with a lower mean bitrate), and so forth.

QUOTE (halb27 @ Apr 26 2013, 23:04)

I personally belong to the paranoid group, and you sound very much like you do too.My advice: use lame3100i, a functional extension. lame3100i has a minimum audio data bitrate feature for instance, something that you obviously want to use but can't get with standard Lame which just allows for defining the mnimum frame bitrate.

Whilst its good that you provide this edited version and everything, I advise against recommending that new users adopt your personal methods simply because they remind you of yourself. Some rudimentary testing might do away with the OPs concerns about bitrate, if not your own, so by recommending your edited version in lieu of such testing, you might be obscuring the truth of the OP actually experiences different bitrates perceptually. In other words, minimal mean bitrates may not be necessary, and I personally doubt they are for the reasons that I implied above about LAME usually being a sufficiently good judge of what an input stream demands.

I personally belong to the paranoid group, and you sound very much like you do too.My advice: use lame3100i, a functional extension. lame3100i has a minimum audio data bitrate feature for instance, something that you obviously want to use but can't get with standard Lame which just allows for defining the mnimum frame bitrate.

Whilst it’s good that you provide this edited version and everything, I advise against recommending that new users adopt your personal methods simply because they remind you of yourself. Some rudimentary testing might do away with the OP’s concerns about bitrate, if not your own, so by recommending your edited version in lieu of such testing, you might be obscuring the truth of the OP actually experiences different bitrates perceptually. In other words, minimal mean bitrates may not be necessary, and I personally doubt they are for the reasons that I implied above about LAME usually being a sufficiently good judge of what an input stream demands.

You as a member of group 1 I described above don't like that, sure. But aren't you a bit on a mission? Why shall I not give an advice to a new member who belongs to group 2 when I belong myself to this group, especially when the advice has very much in focus what the OP wants to do? Yes, my advice here is about my own Lane variant. I have rarely done so before, but when this variant perfectly matches the OP's intention why shouldn't I point to it?As for a listening experience the issue at sec. 3.0 of problem sample eig is a good example. While I can perfectly accept that you (and other members of group1) don't see a problem here because when using standard Lame -V0 the issue is small and issues like these are very rare, I would welcome if you could respect other people's attitude towards things like this, no matter whether they are new or old members.

It all ends up whether we belong to group1 or group2. These groups have different things in mind for what is essential to them, and I think we should simply respect each other. There is no 'right' or 'wrong' here, just different personal needs.

[...] but when this variant perfectly matches the OP's intention why shouldn't I point to it?

I think db1989's point is that you're encouraging the OP to make decisions based on his arbitrary idea of what a suitable bitrate is, when he still has yet to perform any ABX tests. I doubt he would chide you for recommending your LAME extension to someone who has demonstrated an ability to ABX V0 in the default LAME branch, but when the OP has yet to even show that he has ABXed LAME at any level, suggesting that he automatically choose a version that will produce higher bitrates makes little sense and doesn't mesh with TOS8. The OP should determine if even V2 is necessary for him, let alone V0, before he considers moving to your extension.

I would welcome if you could respect other people's attitude towards things like this, no matter whether they are new or old members.

It wasn’t my intention to disrespect anyone or anything. Aleron Ives (thanks!) pretty much hit the nail right on the head in terms of interpreting what I actually meant.

As I implied, I think it’s a good thing that you offer an alternative version targeted to improve specific phenomena. If the OP does sufficiently extensive testing to find the default branch lacking in those areas, then great: he has another option. My point was that people generally should use the default release and settings unless they have a good reason to do otherwise, and recommending a different approach to new users – before they’ve even tried using the official release with recommended settings, never mind testing those objectively – just confuses things further and possibly suggests that the official branch has significant deficiencies that apply in general use.

First it needs to be determined whether the OP really needs a higher mean bitrate than offered by default switches. Enforcing a minimal size on frames doesn’t seem to me to be a useful thing if we assume that LAME is a good judge of instantaneous complexity: clamping the size at 256 kbps/frame or whatever is more likely just to inflate the size of less complex sections. Testing using more conventional settings and evaluating their perceptual performance seems to be more of a priority than worrying about enforcing a minimal mean bitrate; presumably the idea of the latter is based on concerns about quality, which aren’t admissable until testing has shown they exist.

I thank all for the comments and help. I was in no way offended by any of the comments (such as myself crippling the bit-rate by using 256) nor additional stuff that can be added by to lame. Some of the things suggested would be to complex for me, but I have a more of a feel of where to place my settings for lame. The overall "average" of opinion (and yes opinion to me is fine, since that is what I wanted from everyone.) is that ~245 bit resouviour turned on, no minium settings on bit rate. This seems the easist method for a newbie since yes I have no idea what the heck I am doing when it comes to playing around with the settings. I will download foobar is everyone seems to like this program. When I made a comment on "file size" I guess I really shouldn't care since i have plenty of storage space on my mp3 players. But I figured what is the point of wasting space on a 320KCBR when 1.It might be neccessary and 2.Lower bit rate at VBR might actually make the song sound better. Also some say that VBR makes battery life more effiecent. Not sure if this is true or any data to back it up, so that is another reason. I am looking forward to using Foobar when I get back home during the week. THANKS ALL, YOU ALL ROCK

My interpretation of the OP’s references to that and similar descriptions was that MediaMonkey uses the mean bitrate to describe some underlying actual setting of LAME. As I implied when mentioning that before, I don’t much care for that sort of nomenclature due to the confusion that it causes when some files come out with a sizeably higher bitrate, which is of course the point of VBR!